search for: callers

libswfdec/swfdec_as_frame.c | 14 ++ libswfdec/swfdec_as_frame_internal.h | 3 libswfdec/swfdec_as_function.c | 12 -- libswfdec/swfdec_as_interpret.c | 1 libswfdec/swfdec_asbroadcaster.c | 3 test/trace/arguments-5.swf |binary test/trace/arguments-5.swf.trace | 82 +++++++++-------- test/trace/arguments-6.swf |binary

...t/test_bins.sh +++ b/test/test_bins.sh @@ -52,8 +52,8 @@ flac --help 1>/dev/null 2>/dev/null || die "ERROR can't find flac executable" run_flac () { if [ x"$FLAC__TEST_WITH_VALGRIND" = xyes ] ; then - echo "valgrind --leak-check=yes --show-reachable=yes --num-callers=100 flac $*" >>test_bins.valgrind.log - valgrind --leak-check=yes --show-reachable=yes --num-callers=100 --log-fd=4 flac $* 4>>test_bins.valgrind.log + echo "valgrind --leak-check=yes --show-reachable=yes --num-callers=50 flac $*" >>test_bins.valgrind.log + valgr...

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how...

[This feature addresses https://bugs.llvm.org/show_bug.cgi?id=37880 and https://github.com/KSPP/linux/issues/84.] Clang has been ramping up its support of the Linux kernel. We recently added "asm goto with outputs", a long requested feature. We want to continue building our relationship with the Linux community. KSPP is a project to improve security in the Linux kernel, through both

Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various

Hi all, I have the following setup PSTN -> ASTERISK -> IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

# ./testcall testcall.conf Chan 1, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025860' to '013331339767' Chan 2, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025861' to '013331339768' Chan 3, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025862' to

Hello, Users. I loaded module chan_h323.so, chan_vpb.so. I have met a message : "No one is available to answer at this time". I don?t know what I do.. My 'h.323 trace 5' result is : == vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] -- Executing Dial("vpb/1-8", "h323/192.168.1.107") in new stack 1:21:34.936 ThreadID=0x06f2bbb0

Nicol?s Gudi?o <asternic@gmail.com> wrote: >>> Paul, 1.0.5 stable suffers from caller id issues as well, at least for >>> SIP channels. What fixed things for me was swapping in app_dial.c from >>> 1.0.2 stable (didn't try others). You could also just diff app_dial.c >>> between versions to find the problem but I took the lazy way out the >>>

I am trying to implement a change to our Dialplan that will thwart tele-spammers that are calling us with blanked out caller ID. The caller IDs seem to vary between originating callers when they block caller ID. I've seen the following: "anonymous" "" So I'm checking for these. However recently one company seems to be bypassing this, so what I wanted to do was implement some logic that checks for actual numbers in the caller ID. We have a coupl...

Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten => 5150/0,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? -Walter -- Walter Doerr =*= wd@infodn.rmi.de =*=...

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password

Hi, I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Thanks, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ls with spoofed caller ID. I have now changed the dialplan slightly so that the first time people call they are asked to dial 1. After the first call, they are added to a known caller list and get straight through, and any robocalls at that point are blacklisted manually. I have found that most robocallers spoof the Caller ID so rarely call from the same number twice. It means that legitimate callers who cannot dial 1 just have to dial again to get through to the phones - there is a recorded message telling them to dial 1 or call back. I haven't had a robocall since! The hardest thing about this...

I need some advice on "forwarding" arguments to a callee. Suppose I have a function F that is called at the beginning of all other functions in the module. From F I need to access (read) the arguments passed to its immediate caller. Right now I do something like boxing all arguments in the caller inside a struct and passing a pointer to the struct to F, alongside an identifier

Hi, I am following http://xenbits.xen.org/docs/unstable/misc/vtpm.txt, but I''m having some problems when I try to start vtpmmgr-stubdom I''m using Xen 4.3 on Ubuntu 12.04 and I have a physical TPM. The config file for vTPM manager is: kernel="/usr/local/lib/xen/boot/vtpmmgr-stubdom.gz" memory=16 disk=["file:/var/vtpmmgr-stubdom.img,hda,w"]

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I don't see the caller's number in the